The current trend of using Packet Networks (PN) to transport data traditionally carried over circuit switched networks such as the Public Switched Telephone Network (PSTN) creates a need to support the installed-base of terminals attached to the PSTN. FIG. 2 depicts a traditional PSTN configuration as well as the interfacing to a PN substitute network.
The different nature of these two types of networks translates into different characteristics such as bandwidth, delay and loss of information. The differences in characteristics between these networks can affect the terminals that have been designed with PSTN characteristics in mind, if and when the terminals need to communicate over a PN 204. For example, Packet Network connections differ from the traditional telephone networks for which modems, such as modems 218 and 224, were originally designed. Accordingly, many terminals are delay sensitive, and their interaction has been designed on the delay characteristics of the network in use at the time of their inception, typically, the PSTN.
Further, many of the mechanisms designed to compensate for these differences are designed to make human conversation comfortable. For example, in Voice over Internet Protocol (VoIP) network connections, the VoIP processing is typically conducted in the gateways, for example, IP gateways 210 and 212, located at the edge of the IP network. These VoIP processing algorithms are typically optimized as much as possible for handling voice traffic. However, if this optimization is done without any consideration for modems 218 and 224 within user endpoints 230 and 240, any corresponding modem connections may be greatly degraded or may not function over such a VoIP connection, i.e., the voice processing mechanisms will not facilitate reliable modem communications. Further VoIP connections can tend to create jitter that can destroy modem training and end-to-end connectivity.
In a packet network, individual data packets typically traverse the network with different propagation delays, depending on the routes the packets take within the network and the amount of queued data within the network. A VoIP terminating point in gateways 210 and 212 will typically provide a continuous signal by generally adding a large enough throughput delay to absorb the statistical spread in propagation delay. One option for voice traffic is to provide an adaptive jitter buffer that attempts to minimize the throughput delay by adaptively adjusting to changes in the delay statistics over time as the volume of network traffic increases or decreases. However, this adaptive jitter buffer algorithm can tend to degrade modem performance, particularly in echo canceling modulations like V.90 or V.34. For example, as the jitter buffer adapts, changes in the round-trip delay of the associated echo would be observed. Accordingly, the adaptive algorithms in the echo canceller attempt to track these changes in delay. However, for typical echo canceller adaptation rates used by most modems, it is very unlikely that an echo canceller could adapt quickly enough to track any changes in round-trip delay. As such, a change in the round trip delay as small as 2 or 3 ms is likely to cause a modem, such as a V.34 or V.90 modem, to disconnect over a large proportion of line conditions.
In addition, most modems historically have been designed to tolerate up to 1.2 seconds of round trip delay, such as may have been required for the worst case of two satellite hops, i.e., two satellites encountered within the communication path. However, since current technology has rendered unlikely the possibility of ever seeing two satellite hops, as well as with underwater cable systems having a much lower delay becoming more common, modern modem designs are quite likely to have less round trip delay capability than the old 1.2 second design standards. Accordingly, present IP gateways, particularly those with adaptive jitter buffers optimized for voice communications, cannot resolve alone the problems with present modem communications through VoIP networks.
In addition to the network impairment of throughput delay, another impairment consists of packet loss. When packets are lost during a communication session, attempts can be made by the network to minimize the duration of time that the modem receives an invalid signal, such that the modem simply receives a burst of errors during the period of missing data but continues normally thereafter once the data signal resumes. Most modern modems can be somewhat tolerant to impulsive noise, sudden signal hits or drop-outs. Accordingly, if the duration of time that the modem sees invalid data during a lost packet is small enough, the algorithms that give tolerance to, for example, impulsive noise, sudden signal hits, or drop outs, may keep the modem in a stable condition. However, in the event that the period for the loss of data lasts too long, the adaptive algorithms can be thrown off track, and thus, a retraining of the modem is needed.
Thus, a need exist for new methods for providing reliable data communications through packet networks for voiceband communications other than analog voice communications, such as, for example, fax or modem communications. Further, a need exist for a method to improve the throughput delay and packet loss impairments present over IP networks. Additionally, a need exist to facilitate the detection of different types of voiceband traffic, for example, detecting voice versus different modem/fax modulations, and for disabling network voice echo cancellers, adaptive jitter buffers and other such mechanisms implemented in the VoIP algorithms that would degrade modem performance.